Blind source separation method and acoustic signal processing system for improving interference estimation in binaural wiener filtering

ABSTRACT

A method and an acoustic signal processing system for noise reduction of a binaural microphone signal (x 1 , x 2 ) with one target point source and M interfering point sources (n 1 , n 2 , . . . , n M ) as input sources to a left and a right microphone of a binaural microphone system, include:
         filtering a left and a right microphone signal by a Wiener filter to obtain binaural output signals of a target point source, where the Wiener filter is calculated as:       

                 H   W     =     1   -       Φ       (       x     1   ,   n       +     x     2   ,   n         )     ⁢     (       x     1   ,   n       +     x     2   ,   n         )           Φ       (       x   1     +     x   2       )     ⁢     (       x   1     +     x   2       )               ,         
where H W  is the Wiener filter, Φ (x     1,n     +x     2,n     )(x     1,n     +x     2,n     )  is the auto power spectral density of the sum of all of the M interfering point sources components (x1,n, x2,n) contained in the left and right microphone signals and Φ (x     1     +x     2     )(x     1     +x     2     )  is the auto power spectral density of the sum of the left and right microphone signals. Due to the linear-phase property of the calculated Wiener filter, original binaural cues are perfectly preserved not only for the target source but also for the residual interfering sources.

CROSS-REFERENCE TO RELATED APPLICATION

This application claims the priority, under 35 U.S.C. §119, of EuropeanPatent Application EP 090 00 799, filed Jan. 21, 2009; the priorapplication is herewith incorporated by reference in its entirety.

BACKGROUND OF THE INVENTION Field of the Invention

The present invention relates to a method and an acoustic signalprocessing system for noise reduction of a binaural microphone signalwith one target point source and several interfering point sources asinput sources to a left and a right microphone of a binaural microphonesystem. Specifically, the present invention relates to hearing aidsemploying such methods and devices.

In the present document, reference will be made to the followingdocuments:

[BAK05] H. Buchner, R. Aichner, and W. Kellermann. A generalization ofblind source separation algorithms for convolutive mixtures based onsecond-order statistics. IEEE Transactions on Speech and Audio SignalProcessing, January 2005.

[PA02] L. C. Parra and C. V. Alvino. Geometric source separation:Merging convolutive source separation with geometric beamforming. IEEETransactions on Speech and Audio Processing, 10(6):352{362, September2002.

In signal enhancement tasks, adaptive Wiener Filtering is often used tosuppress background noise and interfering sources. Several approachesare proposed for required interference and noise estimates, usuallyexploiting VAD (Voice Activity Detection), and beam-forming, which usesa microphone array with a known geometry. The drawback of VAD is thatthe voice-pause cannot be robustly detected, especially in themulti-speaker environment. The beam-former does not rely on the VAD,nevertheless, it needs a priori information about the source positions.As an alternative method, Blind Source Separation (BSS) was proposed tobe used in speech enhancement, which overcomes the drawbacks mentionedand drastically reduces the number of microphones. However, thelimitation of BSS is that the number of point sources cannot be largerthan the number of microphones, or else BSS is not capable of separatingthe sources.

SUMMARY OF THE INVENTION

It is accordingly an object of the invention to provide a blind sourceseparation method and an acoustic signal processing system for improvinginterference estimation in binaural Wiener filtering, which overcome thehereinafore-mentioned disadvantages of the heretofore-known methods andsystems of this general type and which improve interference estimationin binaural Wiener Filtering in order to effectively suppress backgroundnoise and interfering sources.

With the foregoing and other objects in view there is provided, inaccordance with the invention, a method for noise reduction of abinaural microphone signal. One target point source and M interferingpoint sources are input sources to a left and a right microphone of abinaural microphone system. The method includes the following step:

filtering a left and a right microphone signal by a Wiener filter toobtain binaural output signals of the target point source, where theWiener filter is calculated as:

${H_{W} = {1 - \frac{\Phi_{{({x_{1,n} + x_{2,n}})}{({x_{1,n} + x_{2,n}})}}}{\Phi_{{({x_{1} + x_{2}})}{({x_{1} + x_{2}})}}}}},$where H_(W) is the Wiener filter transfer function Φ_((x) _(1,n) _(+x)_(2,n) _()(x) _(1,n) _(+x) _(2,n) ₎ is the auto power spectral densityof the sum of all of the M interfering point sources componentscontained in the left and right microphone signals and Φ_((x) ₁ _(+x) ₂_()(x) ₁ _(+x) ₂ ₎ is the auto power spectral density of the sum of theleft and right microphone signals.

Due to the linear-phase property of the calculated Wiener filter H_(W),original binaural cues based on signal phase differences are perfectlypreserved not only for the target source but also for the residualinterfering sources.

In accordance with another mode of the invention, the sum of all of theM interfering point sources components contained in the left and rightmicrophone signals is approximated by an output of a Blind SourceSeparation system with the left and right microphone signals as inputsignals.

In accordance with a further mode of the invention, the Blind SourceSeparation includes a Directional Blind Source Separation Algorithm anda Shadow Blind Source Separation algorithm.

With the objects of the invention in view, there is also provided anacoustic signal processing system, including a binaural microphonesystem with a left and a right microphone and a Wiener filter unit fornoise reduction of a binaural microphone signal with one target pointsource and M interfering point sources as input sources to the left andthe right microphone. The Wiener filter unit is calculated as:

${H_{W} = {1 - \frac{\Phi_{{({x_{1,n} + x_{2,n}})}{({x_{1,n} + x_{2,n}})}}}{\Phi_{{({x_{1} + x_{2}})}{({x_{1} + x_{2}})}}}}},$Where Φ_((x) _(1,n) _(+x) _(2,n) _()(x) _(1,n) _(+x) _(2,n) ₎ is theauto power spectral density of the sum of all of the M interfering pointsources components contained in the left and right microphone signalsand Φ_((x) ₁ _(+x) ₂ _()(x) ₁ _(+x) ₂ ₎ is the auto power spectraldensity of the sum of the left and right microphone signals, and theleft microphone signal of the left microphone and the right microphonesignal of the right microphone are filtered by the Wiener filter toobtain binaural output signals of the target point source.

In accordance with another feature of the invention, the acoustic signalprocessing system includes a Blind Source Separation unit, where the sumof all of the M interfering point source components contained in theleft and right microphone signals is approximated by an output of theBlind Source Separation unit with the left and right microphone signalsas input signals.

In accordance with a further feature of the invention, the Blind SourceSeparation unit includes a Directional Blind Source Separation unit anda Shadow Blind Source Separation unit.

In accordance with a concomitant feature of the invention, the left andright microphones of the acoustic signal processing system are locatedin different hearing aids.

Other features which are considered as characteristic for the inventionare set forth in the appended claims.

Although the invention is illustrated and described herein as embodiedin a blind source separation method and an acoustic signal processingsystem for improving interference estimation in binaural Wienerfiltering, it is nevertheless not intended to be limited to the detailsshown, since various modifications and structural changes may be madetherein without departing from the spirit of the invention and withinthe scope and range of equivalents of the claims.

The construction and method of operation of the invention, however,together with additional objects and advantages thereof will be bestunderstood from the following description of specific embodiments whenread in connection with the accompanying drawings.

BRIEF DESCRIPTION OF THE SEVERAL VIEWS OF THE DRAWING

FIG. 1 is a diagrammatic, plan view of a hearing aid according to thestate of the art; and

FIG. 2 is a block diagram of an acoustic scenario being considered and asignal processing system, according to the invention.

DETAILED DESCRIPTION OF THE INVENTION

Referring now to the figures of the drawings in detail and first,particularly, to FIG. 1 thereof, there is seen a hearing aid which isbriefly introduced in the next two paragraphs, since the presentapplication is preferably applicable thereto.

Hearing aids are wearable hearing devices used for supplying aid tohearing impaired persons. In order to comply with numerous individualneeds, different types of hearing aids, such as behind-the-ear hearingaids and in-the-ear hearing aids, e.g. concha hearing aids or hearingaids completely in the canal, are provided. The hearing aids listedabove as examples are worn at or behind the external ear or within theauditory canal. Furthermore, the market also provides bone conductionhearing aids, implantable or vibrotactile hearing aids. In those cases,the affected hearing is stimulated either mechanically or electrically.

In principle, hearing aids have one or more input transducers, anamplifier and an output transducer, as important components. An inputtransducer usually is an acoustic receiver, e.g. a microphone, and/or anelectromagnetic receiver, e.g. an induction coil. The output transducernormally is an electro-acoustic transducer such as a miniature speakeror an electro-mechanical transducer such as a bone conductiontransducer. The amplifier usually is integrated into a signal processingunit. Such a principle structure is shown in FIG. 1 for the example of abehind-the-ear hearing aid. One or more microphones 2 for receivingsound from the surroundings are installed in a hearing aid housing 1 forwearing behind the ear. A signal processing unit 3 is also installed inthe hearing aid housing 1 and processes and amplifies signals from themicrophone. An output signal of the signal processing unit 3 istransmitted to a receiver 4 for outputting an acoustical signal.Optionally, the sound will be transmitted to the ear drum of the hearingaid user through a sound tube fixed with an otoplastic in the auditorycanal. The hearing aid and specifically the signal processing unit 3 aresupplied with electrical power by a battery 5 which is also installed inthe hearing aid housing 1.

In a preferred embodiment of the invention, two hearing aids, one forthe left ear and one for the right ear, have to be used (“binauralsupply”). The two hearing aids can communicate with each other in orderto exchange microphone data.

If the left and right hearing aids include more than one microphone, anypreprocessing that combines the microphone signals into a single signalin each hearing aid can use the invention.

FIG. 2 shows the proposed system which is composed of three majorcomponents A, B and C. The first component A is a linear BSS model in anunderdetermined scenario when more point sources s, n₁, n₂, . . . ,n_(M) than microphones 2 are present. A directional BSS 11 is exploitedto estimate the interfering point sources n₁, n₂, . . . , n_(M) in thesecond component B. Its major advantage is that it can deal with theunderdetermined scenario. In the third component C, an estimatedinterference y₁ is used to calculate a time-varying Wiener filter 14 andthen a binaural enhanced target signal ŝ can be obtained by filteringbinaural microphone signals x₁, x₂ with the calculated Wiener filter 14.Due to the linear-phase property of the calculated Wiener filter 14,original signal-phase-based binaural cues are perfectly preserved notonly for the target source s but also for the residual interferingsources n₁, n₂, . . . n_(M). The application to hearing aids canespecially benefit from this property. A detailed description of theindividual components and experimental results will be presented in thefollowing.

As is illustrated in FIG. 2, one target point source s and M interferingpoint sources n_(m), where m=1, . . . , M are filtered by a linearmultiple-input-multiple-output (MIMO) system 10 before they are pickedup by two microphones 2. Thus, the microphone signals x₁, x₂ can bedescribed in the discrete-time domain by:

$\begin{matrix}{{{x_{j}(k)} = {{{h_{1j}(k)}*{s(k)}} + {\sum\limits_{m = 1}^{M}\;{{h_{{m + 1},j}(k)}*{n_{m}(k)}}}}},} & (1)\end{matrix}$

where “*” represents convolution, h_(ij), where I=1, . . . , M+1 andj=1, 2, denotes a FIR filter model from the I-th source to the j-thmicrophone and x₁, x₂ denote the left and right microphone signal foruse as a binaural microphone signal. Note that in this case the originalsources s, n₁, n₂, . . . , n_(M) are assumed to be point sources so thatthe signal paths can be modeled by FIR filters. In the following, forsimplicity, a time argument k for all signals in the time domain isomitted and time-domain signals are represented by lower-case letters.

The BSS of the component B is desired to find a corresponding demixingsystem W to extract the individual sources from the mixed signals.Output signals of the demixing system y_(i)(k), i=1, 2 are described by:y _(i) =w _(1i) *x ₁ +w _(2i) *x ₂,  (2)

where w_(ji) denotes the demixing filter from the j-th microphone to thei-th output channel.

There are different criteria for the convolutive source separationproposed. They are all based on the assumption that the sources arestatistically independent and can all be used for the invention,although with different effectiveness. In the proposed system, the“TRINICON” criterion for second-order statistics [BAK05] is used as theBSS optimization criterion, where the cost function J_(BSS)(W) aims atreducing the off-diagonal elements of the correlation matrix of the twoBSS outputs:

$\begin{matrix}{{R_{yy}(k)} = {\begin{bmatrix}R_{y_{1}{y_{1}{(k)}}} & R_{y_{1}{y_{2}{(k)}}} \\R_{y_{2}{y_{1}{(k)}}} & R_{y_{2}{y_{2}{(k)}}}\end{bmatrix}.}} & (3)\end{matrix}$

For I=j=2, in each output channel one source can be suppressed by aspatial null. Nevertheless, for the underdetermined scenario no uniquesolution can be achieved. However, in this case Applicants exploit a newapplication of BSS, i.e, its function as a blocking matrix to generatean interference estimate. This can be done by using the Directional BSS11, where a spatial null can be forced to a certain direction forassuring that the source coming from this direction is suppressed wellafter the Directional BSS 11.

The basic theory for the Directional BSS 11 is described in [PA02],where the given demixing matrix is:

$\begin{matrix}{{W = {\begin{bmatrix}w_{11} & w_{21} \\w_{12} & w_{22}\end{bmatrix} = \begin{bmatrix}w_{1}^{T} \\w_{2}^{T}\end{bmatrix}}},} & (4)\end{matrix}$

where w^(T) _(i)=[w_(1i) w_(2i)] (i=1, 2) includes the demixing filterfor the i-th BSS-output channel and is regarded as a beam-former, havinga response which can be constrained to a particular orientation θ, thatdenotes the target source location and is assumed to be known in [PA02].In the proposed system, Applicants designate a “blind” Directional BSSin component B, where θ is not a priori known, but can be detected froma Shadow BSS 12 algorithm as described in the next section. In order toexplain the algorithm, the angle θ is supposed to be given. Thealgorithm for a two-microphone setup is derived as follows:

For a two-element linear array with omni-directional sensors and afar-field source, the array response depends only on the angle θ=θ(q)between the source and the axis of the linear array:

$\begin{matrix}{{{d(q)} = {{d(\theta)} = {{\mathbb{e}}^{{- j}\frac{p}{c}\omega\;\sin\;\theta} = \begin{bmatrix}{\mathbb{e}}^{{- j}\; p_{1}\frac{\omega}{c}\sin\;\theta} \\{\mathbb{e}}^{{- j}\; p_{2}\frac{\omega}{c}\sin\;\theta}\end{bmatrix}}}},} & (5)\end{matrix}$

where d(q) represents the phases and magnitude responses of the sensorsfor a source located at q, p is the vector of the sensor position of thelinear array and c is the sound propagation speed.

The total response for the BSS-output channel i is given by:r=w _(i) ^(T) d(θ).  (6)

Constraining the response to an angle θ is expressed by:

$\begin{matrix}{{{WD}(\theta)} = {\begin{bmatrix}{w_{1}^{T}{d(\theta)}} \\{w_{2}^{T}{d(\theta)}}\end{bmatrix} = {C.}}} & (7)\end{matrix}$

The geometric constraint C is introduced into the cost function:J _(C)(W)=∥WD(θ)−C∥ _(F) ²,  (8)

where ∥A∥_(F) ²=trace{AA^(H)} is the Frobenius norm of the matrix A.

The cost function can be simplified by the following conditions:

1. Only one BSS output channel should be controlled by the geometricconstraint. Without loss of generality the output channel 1 is set to bethe controlled channel. Hence, w^(T) ₂d(θ) is set to be zero in such away that only w^(T) ₁, not w^(T) ₂ is influenced by J_(C)(W).2. In [PA02], the geometric constraint is suggested to be C=I, where Iis the identity matrix, which indicates emphasizing the target sourcelocated at the direction of θ and attenuating other sources. In theproposed system, the target source should be suppressed like in anull-steering beam-forming, i.e. a spatial null is forced to thedirection of the target source. Hence, in this case the geometricconstraint C is equal to the zero-matrix.

Thus, the cost function J_(C)(W) is simplified to be:

$\begin{matrix}{{J_{C}(W)} = {{\begin{matrix}{w_{1}^{T}{d(\theta)}} \\0\end{matrix}}^{2}.}} & (9)\end{matrix}$

Moreover, the BSS cost function J_(BSS)(W) will be expanded by the costfunction J_(C)(W) with the weight η_(c):J(W)=J _(BSS)(W)+η_(C) J _(C)(W).  (10)

In this case, the weight η_(C) is selected to be a constant, typicallyin the range of [0.4, . . . , 0.6] and indicates how important J_(C)(W)is. By forming the gradient of the cost function J(W) with respect tothe demixing filter w*_(j,i) we can obtain the gradient update for W:

$\begin{matrix}\begin{matrix}{\frac{\partial{J(W)}}{\partial W^{*}} = {\frac{\partial{J_{BSS}(W)}}{\partial W^{*}} + {\eta_{C}\frac{\partial{J_{C}(W)}}{\partial W^{*}}}}} \\{= {\frac{\partial{J_{BSS}(W)}}{\partial W^{*}} + {\eta_{C}\begin{bmatrix}\frac{\partial{J_{C}(W)}}{\partial w_{11}^{*}} & \frac{\partial{J_{C}(W)}}{\partial w_{21}^{*}} \\\frac{\partial{J_{C}(W)}}{\partial w_{12}^{*}} & \frac{\partial{J_{C}(W)}}{\partial w_{22}^{*}}\end{bmatrix}}}} \\{= {\frac{\partial{J_{BSS}(W)}}{\partial W^{*}} +}} \\{\eta_{C}\begin{bmatrix}{w_{11} + {w_{12}{\mathbb{e}}^{{- {j{({p_{2} - p_{1}})}}}\frac{\omega}{c}\sin\;\alpha}}} & {{w_{11}{\mathbb{e}}^{{- {j{({p_{2} - p_{1}})}}}\frac{\omega}{c}\sin\;\alpha}} + w_{21}} \\0 & 0\end{bmatrix}}\end{matrix} & (11)\end{matrix}$

Using

$\frac{\partial{J_{C}(W)}}{\partial W^{*}},$only the demixing filters ω₁₁ and ω₂₁ are adapted. In order to preventthe adaptation of ω₁₁, the adaptation is limited to the demixing filterω₂₁:

$\begin{matrix}\begin{matrix}{\frac{\partial(W)}{\partial W^{*}} = {\frac{\partial{J_{BSS}(W)}}{\partial W^{*}} + {\eta_{C}\frac{\partial{J_{C}(W)}}{\partial W^{*}}}}} \\{= {\frac{\partial{J_{BSS}(W)}}{\partial W^{*}} + {\eta_{C}\begin{bmatrix}0 & {{w_{11}{\mathbb{e}}^{{- {j{({p_{2} - p_{1}})}}}\frac{\omega}{c}\sin\;\alpha}} + w_{21}} \\0 & 0\end{bmatrix}}}}\end{matrix} & (12)\end{matrix}$

In the previous section, the angular position θ of the target source isassumed to be known a prior. But in practice, this information isunknown. In order to ascertain that the target source is active and toobtain the geometric information of the target source, a method of‘peak’ detection is used to detect the source activity and positionwhich will be described in the following:

Usually, the BSS adaptation enhances one peak (spatial null) in each BSSchannel in such a way that one source is suppressed by exactly onespatial null, where the position of the peak can be used for the sourcelocalization. Based on this observation, if a source in a definedangular range is active, a peak must appear in the corresponding rangeof the demixing filter impulse responses. Hence, supposing that only onepossibly active source in the target angular range exists, we can detectthe source activity by searching the peak in the range and compare thispeak with a defined threshold to indicate whether the target source isactive or not. Meanwhile, the position of the peak can be converted tothe angular information of the target source. However, once the BSS ofcomponent B is controlled by the geometric constraint, the peak willalways be forced into the position corresponding to the angle θ, even ifthe target source moves from θ to another position. In order to detectthe source location fast and reliably, a shadow BSS 12 without geometricconstraint running in parallel to the main Directional BSS 11 isintroduced, which is constructed to react fast to varying sourcemovement by virtue of its short filter length and periodicalre-initialization. As is shown in FIG. 2, the Shadow BSS 12 detects themovement of the target source and gives its current position to theDirectional BSS 11. In this way, the Directional BSS 11 can apply thegeometric constraint according to the given θ and follows the targetsource movement.

In the underdetermined scenario for a two-microphone setup, one targetpoint source s and M interfering point sources n_(m), m=1, . . . , M arepassed through the mixing matrix. The microphone signals are given byequation (1) and the BSS output signals are given by equation (2). Byapplying the Directional BSS 11, the target source s is well suppressedin one output, e.g. y₁. Thus, the output y₁ of the Directional BSS 11can be approximated by:

$\begin{matrix}{{y_{1} \approx {{w_{11}*x_{1,n}} + {w_{21}*x_{2,n}}} \approx {\sum\limits_{m = 1}^{M}\;{\hat{n}}_{m}}},} & (13)\end{matrix}$

where x_(j,n)(j=1, 2) denotes the sum of all of the interferingcomponents contained in the j-th microphone. If we take a closer look aty₁≈ω₁₁*x_(1,n)+ω₂₁*x_(2,n), actually, it can be regarded as a sum of thefiltered version the interfering components contained in the microphonesignals. Thus, we consider such a Wiener filter, where the input signalis the sum of two microphone signals x₁+x₂, and the desired signal isthe sum of the target source components contained in two microphonesignals x_(1,s)+x_(2,s).

Assuming that all sources are statistically independent, in thefrequency domain, the Wiener filter can be calculated as follows:

$\begin{matrix}\begin{matrix}{H_{W} = \frac{\Phi_{{({x_{1} + x_{2}})}{({x_{1,s} + x_{2,s}})}}}{\Phi_{{({x_{1} + x_{2}})}{({x_{1} + x_{2}})}}}} \\{= \frac{\Phi_{{({x_{1,s} + x_{2,s}})}{({x_{1,s} + x_{2,s}})}}}{\Phi_{{({x_{1} + x_{2}})}{({x_{1} + x_{2}})}}}} \\{{= {1 - \frac{\Phi_{{({x_{1,n} + x_{2,n}})}{({x_{1,n} + x_{2,n}})}}}{\Phi_{{({x_{1} + x_{2}})}{({x_{1} + x_{2}})}}}}},}\end{matrix} & (14)\end{matrix}$

where the frequency argument Ω is omitted, φ_(xy) denotes the crosspower spectral density (PSD) between x and y, and x_(1,n)+x_(2,n)denotes the sum of all of the interfering components contained in twomicrophone signals. As mentioned above, y₁ is regarded as a sum of thefiltered versions of the interfering components contained in themicrophone signals. Thus, y₁ is supposed to be a good approximation forx_(1,n)+x_(2,n). In Applicants' proposed system, Applicants use y₁ asthe interference estimate to calculate the Wiener filter and approximatex_(1,n)+x_(2,n) by y₁:

$\begin{matrix}\begin{matrix}{H_{W} = {1 - \frac{\Phi_{{({x_{1,n} + x_{2,n}})}{({x_{1,n} + x_{2,n}})}}}{\Phi_{{({x_{1} + x_{2}})}{({x_{1} + x_{2}})}}}}} \\{\approx {1 - {\frac{\Phi_{y_{1}y_{1}}}{\Phi_{{({x_{1} + x_{2}})}{({x_{1} + x_{2}})}}}.}}}\end{matrix} & (15)\end{matrix}$

Furthermore, to obtain the binaural outputs of the target sourceŜ=[Ŝ_(L),Ŝ_(R)] both of the left and right microphone signals x₁, x₂will be filtered by the same Wiener filter 14 as shown in FIG. 2. Due tothe linear-phase property of H_(W), in ŝ the binaural cues are perfectlypreserved not only for the target component but also for the residual ofthe interfering components.

The applicability of the proposed system was verified by experiments anda prototype of a binaural hearing aid (computer-based real-timedemonstrator). The experiments have been conducted using speech dataconvolved with the impulse responses of two real rooms with T₆₀=50, 400ms respectively and a sampling frequency of f_(s)=16 kHz. A two-elementmicrophone array with an inter-element spacing of 20 cm was used for therecording. Different speech signals of 10 s duration were playedsimultaneously from 2-4 loudspeakers located at 1.5 m distance from themicrophones. The signals were divided into blocks of length 8192 withsuccessive blocks overlapped by a factor of 2. The length of the mainBSS filter was 1024. The experiments were conducted for 2, 3 and 4active sources individually.

In order to evaluate the performance, the signal-to-interference ratio(SIR) and the logarithm of speech-distortion factors (SDF)

${SDF} = {10\log_{10}\frac{{var}\left\{ {x_{s} - {h_{W}*x_{s}}} \right\}}{{var}\left\{ x_{s} \right\}}}$averaged over both channels was calculated for the total 10 s signal.

TABLE 1 Comparison of SDF and ΔSIR for 2, 3, 4 active sources in twodifferent rooms (measured in dB) number of the sources 2 3 4 anechoicroom SIR_In 5.89 −0.67 −2.36 T₆₀ = 50 ms SDF −14.55 −7.12 −6.64 ΔSIR6.29 6.33 3.05 reverberant room SIR_In 5.09 −0.85 −2.48 T₆₀ = 400 ms SDF−13.60 −5.94 −6.23 ΔSIR 6.13 5.29 3.58

Table 1 shows the performance of the proposed system. It can be seenthat the proposed system can achieve about 6 dB SIR improvement (ΔSIR)for 2 and 3 active sources and 3 dB SIR improvement for 4 activesources. Moreover, in the sound examples the musical tones and theartifacts can hardly be perceived due to the combination of the improvedinterference estimation and corresponding Wiener filtering.

1. A method for noise reduction of a binaural microphone signal (x₁, x₂)with one target point source and M interfering point sources (n₁, n₂, .. . , n_(M)) as input sources to a left and a right microphone of abinaural microphone system, the method comprising the following step:filtering a left and a right microphone signal (x₁, x₂) by a Wienerfilter to obtain binaural output signals (ŝ_(L),ŝ_(R)) of the targetpoint source, where the Wiener filter is calculated as:${H_{W} = {1 - \frac{\Phi_{{({x_{1,n} + x_{2,n}})}{({x_{1,n} + x_{2,n}})}}}{\Phi_{{({x_{1} + x_{2}})}{({x_{1} + x_{2}})}}}}},$where H_(W) is the Wiener filter, Φ_((x) _(1,n) _(+x) _(2,n) _()(x)_(1,n) _(+x) _(2,n) ₎ is an auto power spectral density of a sum of allof the M interfering point sources components (x_(1,n), x_(2,n))contained in the left and right microphone signals (x₁, x₂) and Φ_((x) ₁_(+x) ₂ _()(x) ₁ _(+x) ₂ ₎ is an auto power spectral density of a sum ofleft and right microphone signals (x₁, x₂).
 2. The method according toclaim 1, which further comprises approximating the sum of all of the Minterfering point sources components (x_(1,n), x_(2,n)) contained in theleft and right microphone signals (x₁, x₂) by an output (y₁) of a blindsource separation with the left and right microphone signals (x₁, x₂) asinput signals.
 3. The method according to claim 2, wherein the blindsource separation includes a directional blind source separationalgorithm and a shadow blind source separation algorithm.
 4. An acousticsignal processing system, comprising: a binaural microphone system witha left microphone having a left microphone signal (x₁) and a rightmicrophone having a right microphone signal (x₂); and a Wiener filterunit for noise reduction of a binaural microphone signal (x₁, x₂) withone target point source and M interfering point sources (n₁, n₂, . . . ,n_(M)) as input sources to said left and said right microphones; saidWiener filter unit having an algorithm calculated as:${H_{W} = {1 - \frac{\Phi_{{({x_{1,n} + x_{2,n}})}{({x_{1,n} + x_{2,n}})}}}{\Phi_{{({x_{1} + x_{2}})}{({x_{1} + x_{2}})}}}}},$where Φ_((x) _(1,n) _(+x) _(2,n) _()(x) _(1,n) _(+x) _(2,n) ₎ is an autopower spectral density of a sum of all of the M interfering pointsources components (x_(1,n), x_(2,n)) contained in the left and rightmicrophone signals (x₁, x₂) and Φ_((x) ₁ _(+x) ₂ _()(x) ₁ _(+x) ₂ ₎ isan auto power spectral density of a sum of the left and right microphonesignals (x₁, x₂); and the left microphone signal (x₁) of said leftmicrophone and the right microphone signal (x₂) of said right microphonebeing filtered by said Wiener filter unit to obtain binaural outputsignals (Ŝ_(L),Ŝ_(R)) of the target point source.
 5. The acoustic signalprocessing system according to claim 4, which further comprises a blindsource separation unit having an output (y₁), the sum of all of the Minterfering point sources components (x_(1,n), x_(2,n)) contained in theleft and right microphone signals (x₁, x₂) being approximated by theoutput (y₁) of said blind source separation unit with the left and rightmicrophone signals (x₁, x₂) as input signals.
 6. The acoustic signalprocessing system according to claim 5, wherein said blind sourceseparation unit includes a directional blind source separation unit anda shadow blind source separation unit.
 7. The acoustic signal processingsystem according to claim 4, wherein said left and right microphones arelocated in different hearing aids.
 8. The acoustic signal processingsystem according to claim 5, wherein said left and right microphones arelocated in different hearing aids.
 9. The acoustic signal processingsystem according to claim 6, wherein said left and right microphones arelocated in different hearing aids.